Hi,
I recently discovered http://zaf.github.com/asterisk-mstts/ .
In the page above, it is mentioned you have to subscribe to Microsoft
Translator API on Azure Marketplace.
In Azure Marketplace, I found something called Microsoft Translator.
This API is free within a 2 000 000 characters per
Hi,
I don't want to use INFO for DTMF, I want to transmit it inband.
Therefore I set dtmfmode=inband in the sip.conf general section. But
that didn't make it. As my UAC offered INFO in the Allowed-Header and
Asterisk offered INFO in the 200 OK, DTMF is transmitted via INFO.
Additionally I
Where can I find such ip-lists, please?
Am 05.06.2012 18:40, schrieb Alejandro Imass:
We use complete regional blocks from Wizcraft and blocking at minimum
all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We
block almost anything that is not our actual customer market and screw
On 06-06-12 11:41, Thorsten Göllner wrote:
Where can I find such ip-lists, please?
http://www.ipdeny.com/
Regards,
Patrick
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
Hello Thorsten Göllner,
Am 2012-06-04 11:16:58, hacktest Du folgendes herunter:
What do you want to do? Sending and receiving SMS?
Yes
The OpenSMS Server is located on the Vodafone EasyBox 803 A which is
connected trough the ISDN Cable to my ISDN Card and via the Ethernet to
my Switch
I stricly followed instructions steps 1 and 2 and I'm very to report it works !
Thanks for your detailed answer.
May I post here suggestions that may help others to use this script ?
2012/6/6, Lefteris Zafiris zaf@gmail.com:
On 06/06/2012 10:47 AM, Olivier wrote:
Hi,
I recently
For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it
to zapata.conf it worked.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, May 30, 2012 2:35 PM
To:
On 06/06/2012 09:46 AM, Eric Wieling wrote:
For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it
to zapata.conf it worked.
That means Asterisk (1.4.4.x?) was built against Zaptel, not DAHDI.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber:
We have attempted to upgrade to the latest 1.8 release as well as the
latest asterisk 10 release and are experiencing the same issue with
session-timers and TCP where the session-timers seem to trigger and go
out according to the CLI, but we are not seeing that traffic via tcpdump
on the asterisk
Michelle,
I forwarded your message to the OOH323 maintainer / developer. Here is
his reply.
Vladimir, there is 1.6 asterisk which is unsupported already, you can
recommend upgrade to 1.8 or higher version. I can produce patch for 1.6
but upgrade is better way
Thank you,
Vladimir
On
Vladimir:
Thanks for that! Does the response mean that the fix is already in the latest
Asterisk 1.8 ditribution?
I would prefer not to upgrade this site to 1.8 (since it requires retesting
lots of customer code)...so a patch would be ideal.
Thanks,
Michelle
Owais Ahmad millennium@gmail.com writes:
Hello guys,
I need to be able to throw cdrs on more than one servers at a time. Please
let me know how this can be done.
cdr_adaptive_odbc handles multiple servers. Just define several with
[foo] and [bar] and it Just Works.
/Benny
--
Michelle,
I have re-sent your message to the developer. I will let you know when
I get a reply.
I do believe this fix is incorporated in 1.8, and all ongoing
development is done in 1.8+
Thank you,
Vladimir
On 6/6/2012 10:26 AM, Michelle Dupuis wrote:
Vladimir:
Thanks for that! Does
On Wed, 6 Jun 2012 16:37:01 +0200
Olivier oza_4...@yahoo.fr wrote:
I stricly followed instructions steps 1 and 2 and I'm very to report
it works !
I m glad you got it working. Microsoft really tried it's best to make it
as complicated as possible.
Thanks for your detailed answer.
May I post
The boss wants to move from landline service to VOIP service as a cost-cutting
measure. We have one voice line and one fax line. The telco is billing over
$100 a month for the two. We're using Hylafax for faxing and a PBX for the
voice line.
Our existing PBX is an Intertel Axxess box with the
1) Is there a way I don't know about to get Asterisk to talk to either the
IPRC or the IP Phone+ directly in such a way that gets calls from one to
the other?
Since you've stated that your budget is absolutely zero, I'd have to say no.
It also depends on how the old system connects to the
Hi, I've seen similar questions being asked about this issue but left
unanswered.
A calls B. B attended-transfers the call to C using (polycom, cisco)
phone's transfer button. C does not answer the call. A gets B's voicemail.
However, if B blind-transferred the call to C and C did not answer the
On Jun 6, 2012, at 3:06 PM, Doug Lytle wrote:
1) Is there a way I don't know about to get Asterisk to talk to either the
IPRC or the IP Phone+ directly in such a way that gets calls from one to
the other?
Since you've stated that your budget is absolutely zero, I'd have to say no.
It
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel
Seagraves
Sent: Wednesday, June 06, 2012 2:40 PM
To:
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel
Seagraves
Sent: Wednesday, June 06, 2012 2:40 PM
To:
Daniel Seagraves wrote:
The boss wants to move from landline service to VOIP service as a cost-cutting
measure. We have one voice line and one fax line. The telco is billing over
$100 a month for the two. We're using Hylafax for faxing and a PBX for the
voice line.
Our existing PBX is an
Perhaps another idea (works with extensions with 3 or more digits)
#!/bin/sh
asterisk -rx sip show peers|
grep -vP '(UNKNOWN|Unmonitored)' |
cut -f1 -d/ | grep -P '\d\d\d.*' |
while read PEER
do
echo $PEER
asterisk -rx
On Wed, Jun 6, 2012 at 3:59 PM, John Novack
jnov...@stromberg-carlson.orgwrote:
You have been given an unreasonable charge. No budget but obtain the moon!
You or your boss will live to regret getting into any contract, and with a
company that already has a bad reputation even more so.
One
On 06/06/2012 12:40 PM, Daniel Seagraves wrote:
The boss wants to move from landline service to VOIP service as a cost-cutting
measure. We have one voice line and one fax line. The telco is billing over
$100 a month for the two. We're using Hylafax for faxing and a PBX for the
voice line.
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